The present invention relates to digital subscriber loop applications, and more particularly to multiple voice lines with data over a single shared subscriber loop.
With the popularity of the Internet and the increasing trend of small businesses locating to the home, telephone service providers are experiencing a large and increasing demand for additional voice line service to businesses and homes.
Most central offices (COs) have excess switching capacity for providing additional voice lines to subscribers. Once an additional access line is extended to a subscriber, there is little expense involved in providing voice services and the added line can provide the telephone service provider with incremental revenue generating services.
The conventional approach for providing additional access voice lines to the subscriber is to add analog subscriber loops by laying additional copper lines to, and changing or adding lightning protection devices at, the subscriber premise. The subscriber loop is the two-wire copper transmission and signaling path between a telephone subscriber""s terminal equipment and the serving central office or another piece of terminal equipment. However, the time and expense involved in this approach can be considerable, greatly increasing the time to recoup a return on investment.
A problem with any analog subscriber loop based signaling system, from a transmission perspective, is loss and impairment of the signal. This can be caused by physical conditions, such as bridge taps, gauge changes, line length, insulation, age, and environmental cable damage, or due to interference from external sources such as impulse noise and cross talk. Signal degradation typically manifests as noise, loss, distortion, and interference.
Another problem with the conventional approach is that analog loops are typically used with standard modems which use baseband POTS (Plain Old Telephone Service) voice frequency spectrum (0-4 kHz) to transmit information, and cannot exceed transmission power levels as dictated by the FCC due to cable pair crosstalk effects. The effect of the current FCC rules is to restrict the output of service providers"" modems to download speeds of 53 kbps and upload speeds of 31.2 kbps. Actual speeds may vary depending on line conditions, but cannot exceed these maximums.
Frequency Division Multiplexing (FDM) is one technique for providing additional voice lines over a subscriber loop that does not require laying additional copper lines. This approach uses a frequency spectrum that is spectrally isolated from that used by baseband POTS, thus allowing additional 4 kHz analog POTS channels on higher frequency carrier signals to use the same two-wire subscriber loop. Such passband analog carrier techniques tend to amplify the loss and impairments analog loops typically suffer.
A technique that uses FDM is Digital Added Main Line (DAML). At the CO, a DAML modem is presented with two or more subscriber loop analog voice signals. These analog voice signals are converted by the modem to a digital line code format and transmitted over a single subscriber loop to another DAML modem located at or near the customer premise. The customer premise DAML modem decodes the line and presents the subscriber with two or more two-wire connections corresponding to the subscriber loop connections to the DAML modem at the CO. The digital line codes can take a number of forms, the most common of which are Amplitude, Phase and Frequency Shift Keying, 2-Binary-1-Quaternary, Carrierless Amplitude Phase Modulation, and Quadrature Amplitude Phase Modulation. A problem with this approach is that the D/A/D conversion at the CO of the pulse code modulation (PCM) digital signal to an analog loop signal back to the DAML digital signal can cause degradation of the signal through such effects as quantization errors and phase distortion.
Another technique used to transport multiple voice lines in a digital fashion over the subscriber loop is Integrated Services Digital Network (ISDN). This is a direct digital, multiple voice/data channel system that also includes a signaling channel. However, ISDN requires changes in equipment, administration and maintenance at the switching system.
Another approach involves transmitting voice packets over a data network which can include subscriber loops. The better known implementations of this approach are Voice Over IP (VOIP), Voice Over ATM (VOATM), and Voice Over Frame Relay (VOFR).
VOIP applications are typically deployed throughout a campus environment, using CAT 5 wiring or fiber as described in standards publication EIA/TIA-570-91, xe2x80x9cResidential and Light Commercial Telecommunications Wiring,xe2x80x9d Electronic Industries Alliance/Telecommunications Industry Association, June 1991, to each terminal and connected through a common switching fabric such as Ethernet, ATM or a hybrid system. In addition, calls can bridge to the Internet from the campus environment, or Intranet, via gateways such as routers or Layer 3 switching systems.
In some applications, a desktop computer or other device acts as the VOIP enabled terminal used to support remote communications consistent with ITU-T standards publication H.323, xe2x80x9cPacket Based Multimedia Communications Systems,xe2x80x9d International Telecommunications Union (ITU), Feb 1998. Such systems, typically employ Digital Signal Processors (DSPs) to provide compression of voice IP packets at the desktop which are then forwarded to other stations on the local Intranet or on through the Internet to remote stations. VOATM and VOFR are other packet techniques used to transport voice and interwork3 with the Public Switched Telephone Network (PSTN).
Subscriber loops can extend the reach of a WAN network for VOIP applications using xDSL signaling and transmission techniques. xDSL technologies enable bandwidth to the premise that may co-exist with baseband POTS service. ISDN can also provide bandwidth to the home that connects to a packet network through which it provides voice services. IP packets, ATM Cells, or other frame formats can be transported over subscriber loops using ISDN or xDSL technologies such as ADSL and HDSL.
However, voice and data have different requirements for network services. Voice transmission requires only a small amount of bandwidth, but that bandwidth must be available on a dedicated or continuous basis with very little delay, delay variation, or loss. Even delays in the millisecond range can give rise to noticeable echoes or gaps in the conversation. For example, delays introduced by routers and gateways can have adverse affects on voice.
Packetized speech belongs to the category of realtime data traffic, and as such has stringent delivery requirements with respect to loss and error. In packetized speech, the end-to-end average network delivery time must be small, and the end-to-end variation of the delivery time, including losses, must be small.
In voice transmission, the overall delay should not exceed 200 ms, which is the delay that has been accepted as commercially acceptable. 100-200 ms is the typical goal. At around 800 ms, the delay impedes normal telephonic conversation. Normally, a delay of 200-800 ms is conditionally acceptable for a short portion of the conversation when such occurrences are rare and far apart.
In traditional voice networks, the round trip delay is about 20-30 ms. Voice delays in frame relay networks, can be around 125-200 ms. In Ethernet networks carrying TCP/IP packets, the delay can vary widely depending on traffic loads. Due to the inherent realtime deficiencies of shared data networking technologies, the above issues represent serious challenges for the transmission of voice over typical campus networking environments extended to the premise.
In addition, ATM as a standard still lacks support for voice compression, silence suppression, idle channel cell suppression and signaling support including translation of voice signaling to switched virtual connection ATM signaling.
Further, ATM trunking for narrowband services, such as voice, introduces some additional delay to that encountered naturally over an ATM network due to buffering to accommodate cell delay variation introduced by the ATM network and cell assembly/disassembly delay.
Accordingly, it is an object of the present invention to provide a system that provides additional voice and data lines over a subscriber loop that is shared with POTS.
Another object of the present invention is to provide such a system such that any in-band signaling in support of advanced telephony features is supported.
Another object of the present invention is to provide such a system such that there is no discernible degradation of the reconstructed voice signal.
Another object of the present invention is to provide such a system such that voice traffic has priority over data traffic.
Another object of the present invention is to provide such a system such that the analog modem transmission speed of the line is not capped at the FCC mandated limit of 53.3 kbps for analog lines.
Another object of the present invention is to provide such a system that avoids impairments associated with extra D/A conversions.
Another object of the present invention is to provide such a system using less expensive components commonly found in the data network environment.
Another object of the present invention is to provide such a system that is customer installable at the customer""s premise.
Another object of the present invention is to provide such a system that can be incrementally implemented in the network to initially provide voice services without the need to establish a separate data transport infrastructure apart from the current CO switch.
The present invention is a system that provides additional voice lines over a single two-wire subscriber loop while retaining POTS service to the customer premise over the loop. The system includes a digital modem at the CO connected over the subscriber loop to another digital modem at the customer premise. The CO modem connects to the PSTN over a direct digital interface to the CO switch. This eliminates the extra D/A conversion found in most prior art systems. These D/A conversions are a key source of signal degradation, and the elimination of even one such conversion will provide for an improved signal over the prior art systems. The CO modem can also connect to a data network over a packet interface, such as Ethernet.
The CO modem receives voice data from the CO switch over the digital trunk interface in PCM format clocked to the network timing reference signal. The PCM voice samples are then packetized by the CO modem and transmitted to the customer premise modem using a suitable digital modulation line code. A table mapping digital trunk interface timeslots to telephone line addresses on the customer premise modem is maintained in the CO modem. A timing reference signal synchronized with the network timing reference signal is also transmitted from the CO modem to the customer premise modem. In the upstream direction, PCM sample voice packets are received from the customer premise modem, are depacketized and presented to the CO switch in PCM format clocked to the network timing reference signal over the digital trunk interface.
At the customer premise, one or more telecommunication devices are connected to the digital modem into a voice interface. These telecommunication devices can include telephone instruments, such as traditional telephones, xe2x80x9csmart phones,xe2x80x9d analog modems, or facsimile (FAX) machines.
PCM sample voice packets transmitted from the CO modem are received by the customer premise modem, converted to analog voice band signals using the timing reference signal as a clock, and transmitted to the addressed telephone instrument. In the upstream direction, analog voice band signals from the telephone instruments are converted to PCM samples using the timing reference signal as a clock, packetized into voice packets and transmitted by the customer premise modem to the CO modem over the subscriber loop using the digital modulation line code.
At the customer premise, data devices can be connected to the customer premise modem over a data interface. The data devices can be any source of data packets, for example a computer, bridge, router, or hub, behind which a number of individual computers can reside. Data packets received by the customer premise modem from the CO modem are routed to the addressed data device. In the upstream direction, data packets generated by data devices are transmitted over the subscriber loop using the digital modulation line code to the CO modem.
In this system, the only non-digital transmission stage in the architecture is the analog loop at the subscriber premise connecting the telecommunication devices to the premise modem. This distance is typically only a few feet. This digital architecture greatly reduces the loss and impairments found in conventional systems resulting from the analog path between the telephone instrument and the CO switch.
In the present invention, both voice and data packets are transmitted over the subscriber loop to the CO. However, a key object of the invention is preserving the intelligibility of voice communications. This is accomplished by differentiating between voice and data at both the CO and premise modem, via voice and data interfaces, and using a packet transmission priority scheme that gives voice packets a higher priority than data packets. U.S. Pat. No. 5,692,035 to O""Mahoney et al., and xe2x80x9cA Quality of Service Architecturexe2x80x9d by Campbell, Coulson, Hutchison, Computer Communication Review, Vol. 24, No. 2, April 1994, describe examples of such packet based systems giving transmission priority to certain packet classes. This packet transmission priority scheme ensures that voice is transported in the presence of data with a sufficiently low latency so as to ensure speech intelligibility, and overcomes the speech intelligibility issues commonly associated with VOIP, VOATM and VOFR.